Not able to call from softphone

Yash3120 · March 16, 2021, 9:53am

I set up a new machine and install asterisk 16.16.2 on that, and use the same pjsip configuration from my last machine which had the same os configuration and it worked well.
I try to do the same with my new machine but I am facing an error in the account registering, I can register the user but when I dial call I got error like below,

res_pjsip_session.c:934 handle_incoming_sdp:  1102: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)

I don’t know what’s wrong I did.

My pjsip.conf

;================================ TRANSPORTS ==
; Our primary transport definition for UDP communication behind NAT.
[transport-udp-nat]
type = transport
protocol = udp
bind = 0.0.0.0

[endpoint-internal-d70](!)
type = endpoint
context = Long-Distance
allow = !all,ulaw
direct_media = no
trust_id_outbound = yes
device_state_busy_at = 1
dtmf_mode = rfc4733
use_avpf = yes
rtp_symmetric = yes

[auth-userpass](!)
type = auth
auth_type = userpass

[aor-single-reg](!)
type = aor
max_contacts = 1

[1101](endpoint-internal-d70)
auth = 1101
aors = 1101
callerid = Maria Berny <1101>

[1101](auth-userpass)
password = yash1234
username = 1234

[1101](aor-single-reg)
mailboxes = 1101@example

[1102](endpoint-internal-d70)
auth = 1102
aors = 1102
callerid = Yash Mistry <1102>

[1102](auth-userpass)
password = yash5678
username = 5678

[1102](aor-single-reg)
mailboxes = 1102@example

[1103](endpoint-internal-d70)
auth = 1103
aors = 1103
callerid = Penelope Bronte <1103>

[1103](auth-userpass)
password = yash9012
username = 9012

[1103](aor-single-reg)
mailboxes = 1103@example

jcolp · March 16, 2021, 10:14am

You would need to show the actual SIP traffic using “pjsip set logger on”. It appears as though there is no common codec.

Yash3120 · March 16, 2021, 10:50am

mtech*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (1149 bytes) from UDP:192.168.1.7:46287 --->
INVITE sip:1101@192.168.1.190:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:46287;branch=z9hG4bK.FQjAWO8D8;rport
From: <sip:1102@192.168.1.190:5060>;tag=euNq4-x51
To: sip:1101@192.168.1.190:5060
CSeq: 20 INVITE
Call-ID: 7U~~wV6bY8
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 516
Contact: <sip:1102@192.168.1.7:46287;transport=udp>;+sip.instance="<urn:uuid:0361025b-b2e8-0029-b75e-5633b6652352>"
User-Agent: LinphoneAndroid/4.3.1 (Redmi Note 5) LinphoneSDK/4.4.2 (tags/4.4.2^0)

v=0
o=1102 1594 2611 IN IP4 192.168.1.7
s=Talk
c=IN IP4 192.168.1.7
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7079 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (454 bytes) to UDP:192.168.1.7:46287 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.7:46287;rport=46287;received=192.168.1.7;branch=z9hG4bK.FQjAWO8D8
Call-ID: 7U~~wV6bY8
From: <sip:1102@192.168.1.190>;tag=euNq4-x51
To: <sip:1101@192.168.1.190>;tag=z9hG4bK.FQjAWO8D8
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1615891812/d3e37103bf00f724753840a2614a2ed0",opaque="7f699dc505ea32d9",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.16.2
Content-Length:  0


<--- Received SIP request (388 bytes) from UDP:192.168.1.7:46287 --->
ACK sip:1101@192.168.1.190:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:46287;branch=z9hG4bK.FQjAWO8D8;rport
Call-ID: 7U~~wV6bY8
From: <sip:1102@192.168.1.190:5060>;tag=euNq4-x51
To: <sip:1101@192.168.1.190:5060>;tag=z9hG4bK.FQjAWO8D8
Contact: <sip:1102@192.168.1.7:46287;transport=udp>;+sip.instance="<urn:uuid:0361025b-b2e8-0029-b75e-5633b6652352>"
Max-Forwards: 70
CSeq: 20 ACK


<--- Received SIP request (1433 bytes) from UDP:192.168.1.7:46287 --->
INVITE sip:1101@192.168.1.190:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:46287;branch=z9hG4bK.w6tCoNNmN;rport
From: <sip:1102@192.168.1.190:5060>;tag=euNq4-x51
To: sip:1101@192.168.1.190:5060
CSeq: 21 INVITE
Call-ID: 7U~~wV6bY8
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 516
Contact: <sip:1102@192.168.1.7:46287;transport=udp>;+sip.instance="<urn:uuid:0361025b-b2e8-0029-b75e-5633b6652352>"
User-Agent: LinphoneAndroid/4.3.1 (Redmi Note 5) LinphoneSDK/4.4.2 (tags/4.4.2^0)
Authorization:  Digest realm="asterisk", nonce="1615891812/d3e37103bf00f724753840a2614a2ed0", algorithm=md5, opaque="7f699dc505ea32d9", username="5678",  uri="sip:1101@192.168.1.190:5060", response="d68f3d6851a0109d0099cfbedd34a114", cnonce="kXIOdf9hGIZKZGIR", nc=00000001, qop=auth

v=0
o=1102 1594 2611 IN IP4 192.168.1.7
s=Talk
c=IN IP4 192.168.1.7
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7079 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (280 bytes) to UDP:192.168.1.7:46287 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.7:46287;rport=46287;received=192.168.1.7;branch=z9hG4bK.w6tCoNNmN
Call-ID: 7U~~wV6bY8
From: <sip:1102@192.168.1.190>;tag=euNq4-x51
To: <sip:1101@192.168.1.190>
CSeq: 21 INVITE
Server: Asterisk PBX 16.16.2
Content-Length:  0


[Mar 16 16:20:12] ERROR[14411]: res_pjsip_session.c:934 handle_incoming_sdp:  1102: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
<--- Transmitting SIP response (334 bytes) to UDP:192.168.1.7:46287 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.7:46287;rport=46287;received=192.168.1.7;branch=z9hG4bK.w6tCoNNmN
Call-ID: 7U~~wV6bY8
From: <sip:1102@192.168.1.190>;tag=euNq4-x51
To: <sip:1101@192.168.1.190>;tag=dd08b828-c349-432b-a9ca-a6d9570dc74e
CSeq: 21 INVITE
Server: Asterisk PBX 16.16.2
Content-Length:  0


<--- Received SIP request (407 bytes) from UDP:192.168.1.7:46287 --->
ACK sip:1101@192.168.1.190:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:46287;branch=z9hG4bK.w6tCoNNmN;rport
Call-ID: 7U~~wV6bY8
From: <sip:1102@192.168.1.190:5060>;tag=euNq4-x51
To: <sip:1101@192.168.1.190:5060>;tag=dd08b828-c349-432b-a9ca-a6d9570dc74e
Contact: <sip:1102@192.168.1.7:46287;transport=udp>;+sip.instance="<urn:uuid:0361025b-b2e8-0029-b75e-5633b6652352>"
Max-Forwards: 70
CSeq: 21 ACK

jcolp · March 16, 2021, 10:54am

You have “use_avpf” set to “yes” but the client is not using AVPF. You can either disable it or set the " media_use_received_transport" option to “yes” on the endpoint.

system · April 15, 2021, 10:54am

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Not able to call from softphone

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